(or ~500 chars max on LOW_MEMORY) to 80 chars max). This will allow more
channels to be used in a single hint.
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r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) | 5 lines
The passed extension may not be the same in the list as the current entry,
because we strip spaces when copying the extension into the structure.
Therefore, use the copied item to place the item into the list.
(found by lmadsen on -dev, fixed by me)
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Reported by: smurfix
This reverts a change I made in 116297. At the time it seemed the change was required to solve an issue with attempting a transfer but then letting it timeout without dialing any digits. However, I didn't realize that having an empty extension was possible. I'm removing the immediate return that was added in pbx_find_extension if the extension is null.
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r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) | 18 lines
It turns out that the 0x0XX00 codes being returned for
N, X, and Z are off by one, as per conversation with
jsmith on #asterisk-dev; he was teaching a class
and disconcerted that this published rule was not
being followed, with patterns _NXX, _[1-8]22 and
_[2-9]22... and NXX was winning, but [1-8] should
have been.
This change, tested on these 3 patterns now
picks the proper one.
However, this change may surprise users who
set up dialplans based on previous behavior,
which has been there for what, 2 and half
years or so now.
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- Add CLI aliases module to asterisk.
- Remove all deprecated CLI commands from the code
Initial work done by file.
Junk-Y and lmadsen did a lot of work and testing to
get the list of deprecated commands into the configuration file.
Deprecated CLI commands are now handled by this new module,
see cli_aliases.conf for more info about that.
ok russellb@ via reviewboard
(closes issue #13735)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
container. The OBJ_MULTIPLE flag must be provided here. Otherwise, this would
only remove a single object.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Functions are printed without parenthesis like: FUNTION
Applications are printed with parenthesis like: AppName()
Cli commands are printed like: 'core show application'
The other type of references are printed as they are inside the <ref> tag.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format. Currently, a new format is available for
applications and dialplan functions. A good number of conversions to the new format
are also included.
For more information, see the following message to asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html
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the trie info when they do 'dialplan show ...'
(even with debug set to non-zero); so I set up a
'dialplan debug [context]' cli command instead,
to explicitly show just the trie info. I even
added an extension_exists() call to make sure the
trie info is built. I moved the explanatory header
to above the extension loop to ensure it only prints
once. And it will do this now, whether debug is set
or not.
I removed the trie printing from the 'dialplan show'
command entirely.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: nickpeirson
Patches:
pbx.c.patch uploaded by nickpeirson (license 579)
replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
Tested by: nickpeirson, murf
1. replaced all refs to bzero and bcopy to memset and memmove instead.
2. added a note to the CODING-GUIDELINES
3. add two macros to asterisk.h to prevent bzero, bcopy from creeping
back into the source
4. removed bzero from configure, configure.ac, autoconfig.h.in
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r144677 | murf | 2008-09-26 11:47:13 -0600 (Fri, 26 Sep 2008) | 12 lines
(closes issue #13563)
Reported by: mnicholson
Patches:
found1.diff uploaded by mnicholson (license 96)
This patch was mainly meant to apply to trunk and 1.6.x,
but I'm applying it to 1.4 also, which should be a perfectly
harmless fix to the vast majority of users who are not using
external switches, but the few who might be affected
will not have to go to the pain of filing a bug report.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: nickpeirson
The user attached a patch, but the license is not yet
recorded. I took the liberty of finding and replacing
ALL index() calls with strchr() calls, and that
involves more than just main/pbx.c;
chan_oss, app_playback, func_cut also had calls
to index(), and I changed them out. 1.4 had no
references to index() at all.
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r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines
Tested by: sergee, murf, chris-mac, andrew, KNK
This is a "second attempt" to restore the previous "endbeforeh" behavior
in 1.4 and up. In order to capture information concerning all the
legs of transfers in all their infinite combinations, I was forced
to this particular solution by a chain of logical necessities, the
first being that I was not allowed to rewrite the CDR mechanism from
the ground up!
This change basically leaves the original machinery alone, which allows
IVR and local channel type situations to generate CDR's as normal, but
a channel flag can be set to suppress the normal running of the h exten.
That flag would be set by the code that runs the h exten from the
ast_bridge_call routine, to prevent the h exten from being run twice.
Also, a flag in the ast_bridge_config struct passed into ast_bridge_call
can be used to suppress the running of the h exten in that routine. This
would happen, for instance, if you use the 'g' option in the Dial app.
Running this routine 'early' allows not only the CDR() func to be used
in the h extension for reading CDR variables, but also allows them to
be modified before the CDR is posted to the backends.
While I dearly hope that this patch overcomes all problems, and
introduces no new problems, reality suggests that surely someone
will have problems. In this case, please re-open 13251 (or 13289),
and we'll see if we can't fix any remaining issues.
** trunk note: some code to suppress the h exten being run
from app_queue was added; for the 'continue' option available
only in trunk/1.6.x.
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r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008) | 7 lines
When doing an async goto, detect if the channel is already in the middle of a
masquerade. This can happen when chan_local is trying to optimize itself out.
If this happens, fail the async goto instead of bursting into flames.
(closes issue #13435)
Reported by: geoff2010
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of changes to the Set and MSet apps, so people aren't
so shocked and surprised when they upgrade from
1.4 to 1.6.
Also, for the sake of those upgrading from 1.4 to
1.6 with AEL, I provide automatic support for the
"old" way of using Set(), that still does the
exact same old thing with quotes and backslashes
and so on as 1.4 did, by having AEL compile in the
use of MSet() instead of Set(), everywhere it inserts
this code.
But, if the app_set var is set to 1.6 or higher,
it uses the "new", non-evaluative Set().
This only usually happens if the user manually
inserts this into the asterisk.conf file, or runs
the "make samples" command.
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r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) | 14 lines
(closes issue #13409)
Reported by: tomaso
Patches:
asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license 564)
I basically spent the day, verifying that this patch
solves the problem, and doesn't hurt in non-problem
cases. Why valgrind did not plainly reveal this leak
absolutely mystifies and stuns me.
Many, many thanks to tomaso for finding and providing the fix.
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r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9 lines
This patch reverts the changes made via 139347, and 139635, as users
are seeing adverse difference.
I will un-close 13251.
Back to the drawing board/ concept/ beginning/ whatever!
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r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) | 47 lines
(closes issue #13251)
Reported by: sergee
Tested by: murf
THis is a bold move for a static release fix, but I wouldn't have
made it if I didn't feel confident (at least a *bit* confident)
that it wouldn't mess everyone up.
The reasoning goes something like this:
1. We simply cannot do anything with CDR's at the current point
(in pbx.c, after the __ast_pbx_run loop). It's way too late to
have any affect on the CDRs. The CDR is already posted and gone,
and the remnants have been cleared.
2. I was very much afraid that moving the running of the 'h'
extension down into the bridge code (where it would be now
practical to do it), would result in a lot more calls to the
'h' exten, so I implemented it as another exten under another
name, but found, to my pleasant surprise, that there was a
1:1 correspondence to the running of the 'h' exten in the
pbx_run loop, and the new spot at the end of the bridge.
So, I ifdef'd out the current 'h' loop, and moved it into
the bridge code. The only difference I can see is the stuff
about the AST_PBX_KEEPALIVE, and hopefully, if this
is still an important decision point, I can replicate it
if there are complaints. To be perfectly honest,
the KEEPALIVE situation is not totally clear to me,
and how it relates to a post-bridge situation is less
clear. I suspect the users will point out everything
in total clarity if this steps on anyone's toes!
3. I temporarily swap the bridge_cdr into the channel
before running the 'h' exten, which makes it possible
for users to edit the cdr before it goes out the door.
And, of course, with the endbeforehexten config var set,
the users can also get at the billsec/duration vals.
After the h exten finishes, the cdr is swapped back
and processing continues as normal.
Please, all who deal with CDR's, please test this version
of Asterisk, and file bug reports as appropriate!
........
I also made a little fix to the app_dial's 'e' option,
that is related to my updates.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to the Set app in trunk/1.6.x, as they come from the 1.4 world. They are only bitten if
they write their AEL dialplan in the 1.4 world, and then carry it over to a trunk/1.6.x
installation where a "make samples" was executed, or where they hand-edited the
asterisk.conf file and added the [compat] category with app_set = 1.6 (or higher).
(this commit does not totally solve 13249, at least not yet)
The change involves issueing a single warning while the AEL file is loading, if:
1. app_set is present in the config file, and set to 1.6 or higher.
2. there are double quotes in an assignment statement (eg x = "hi there";)
3. the warning was not already issued.
The standalone app, aelparse, does not (yet) issue this warning. I'd have to
have it read in the asterisk.conf file, and that's a bit of hassle. I'll add
it if users request it, tho.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138815 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: falves11
Tested by: murf
falves11 ==
The changes I introduce here seem to clear up the problem
for me. However, if they do not for you, please reopen this
bug, and we'll keep digging.
The root of this problem seems to be a subtle memory corruption
introduced when creating an extension with an empty extension
name. While valgrind cannot detect it outside of DEBUG_MALLOC
mode, when compiled with DEBUG_MALLOC, this is certain death.
The code in main/features.c is a puzzle to me. On the initial
module load, the code is attempting to add the parking extension
before the features.conf file has even been opened!
I just wrapped the offending call with an if() that will not
try to add the extension if the extension name is empty. THis
seems to solve the corruption, and let the "memory show allocations"
work as one would expect.
But, really, adding an extension with an empty name is a seriously
bad thing to allow, as it will mess up all the pattern matching
algorithms, etc. So, I added a statement to the add_extension2 code to return
a -1 if this is attempted.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: murf
Tested by: murf
For: J. Geis
The 'data' field in the ast_exten struct was being
'moved' from the current dialplan to the replacement
dialplan. This was not good, as the current dialplan
could have problems in the time between the change
and when the new dialplan is swapped in.
So, I modified the merge_and_delete code to strdup
the 'data' field (the args to the app call), and
then it's freed as normal.
I improved a few messages; I added code to limit
the number of calls to the context_merge_incls_swits_igps_other_registrars()
to one per context. I don't think having it called
multiple times per context was doing anything bad,
but it was inefficient.
I hope this fixes the problems Mr. Geiss was noting in
asterisk-users, see
http://lists.digium.com/pipermail/asterisk-users/2008-July/215634.html
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Reported by: mnicholson
Spent most of the day on this bug, and the
solution was so simple. Just had to find and
understand the problem.
The problem was, that the routine to copy
the existing switches, includes, and ignorepats
from the old context to the new one, wasn't
getting called when the context is already
existent. (In other words, if AEL is adding
a new context to the mix, they get copied,
but if pbx_config already defined a context,
then the copy wasn't happening. This made
no sense, so I moved the call to copy the
includes & etc, no matter the case.
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r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines
astman_send_error does not need a newline appended -- the API takes care of
that for us.
(closes issue #13068)
Reported by: gknispel_proformatique
Patches:
asterisk_1_4_astman_send.patch uploaded by gknispel (license 261)
asterisk_trunk_astman_send.patch uploaded by gknispel (license 261)
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Reported by: eliel
OK, now the context registrar slot is strdup'd. It is freed
on destruction. I don't see the need to do this with all
the structs' registrar fields, but if some wild case proves
they should also be handled this way, then we can
put in the extra work at that time.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: eliel
Tested by: murf
(closes issue #12960)
Reported by: mnicholson
In this 'omnibus' fix, I **think** I solved both
the problem in 13041, where unloading pbx_ael.so
caused crashes, or incomplete removal of previous
registrar'ed entries. And I added code to completely
remove all includes, switches, and ignorepats that
had a matching registrar entry, which should
appease 12960.
I also added a lot of seemingly useless brackets
around single statement if's, which helped debug
so much that I'm leaving them there.
I added a routine to check the correlation between
the extension tree lists and the hashtab
tables. It can be amazingly helpful when you have
lots of dialplan stuff, and need to narrow
down where a problem is occurring. It's ifdef'd
out by default.
I cleaned up the code around the new CIDmatch code.
It was leaving hanging extens with bad ptrs, getting confused
over which objects to remove, etc. I tightened
up the code and changed the call to remove_exten
in the merge_and_delete code.
I added more conditions to check for empty context
worthy of deletion. It's not empty if there are
any includes, switches, or ignorepats present.
If I've missed anything, please re-open this bug,
and be prepared to supply example dialplan code.
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AMI commands can display that a channel is under control of an AGI.
Work inspired by work at customer site, but paid for by Edvina AB
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